Internet protocol (IP) networks, and the like, for providing data communication among a plurality of computers are well-known. Such networks facilitate the transfer of data files, audio information and video information, as well as any other information that may be represented in binary form among the plurality of computers.
Networks can be conveniently divided into two broad categories, based upon their size. A local area network (LAN) is a group of computers which is connected so as to facilitate the sharing of applications, data and peripherals. Local area networks are generally confined to a single building or a small group of buildings.
A wide area network (WAN) is made up of a plurality of LANs which is connected together so as to facilitate communication therebetween. A WAN may cover a city, a state, a country or even be international in scope. The Internet is an example of a WAN that includes more than 2,000 separate packet-switched networks that are located all over the world.
The popularity of networks, such as the Internet, has increased the desire for additional network services, such as network telephony. The vast, high bandwidth network provides an ideal medium for audio communications. The nature of such telephone devices is to process voice signals that might come in over the network, typically as digital packets of information, or the like. To process such signals, various computing and processing devices are used, typically in the form of integrated circuit configurations. These devices are often capable of handling multiple channels of information. Alternatively, multiple channels might be processed using multiple devices distributed across a network.
For telephony and voice applications, packet-based networks are becoming more widely used. In the past, it was often difficult to guarantee sufficient bandwidth or a certain “quality of service” that might be needed to accommodate real-time voice signals (that are broken into packets). However, known solutions have been developed which address various problems associated with transmitting voice over packet networks, including, for instance, voice over IP (Internet Protocol), voice over ATM (asynchronous transfer mode), and voice over frame relay.
Prior art solutions include voice coders (i.e., particularized hardware and software devices) that can be applied to packet-based voice systems. Voice coders are configured to process and encode incoming voice signals at certain data rates. In general, if a higher data rate is used (i.e., more voice samples are taken), then the voice coder does not need to be as complex. In other words, fewer hardware components might be employed, or the software associated with the device might be less complex. Conversely, if a lower data rate is used, the voice coder is generally more complex since more compression needs to be performed on the signal.
This data rate can be varied, depending upon the input signal or source being supplied to the device. For such devices, the term variable bit rate (VBR) has been defined as follows: A VBR encoder outputs a bit stream which may have a variable number of bits in successive frames. That is, each frame may contain a different number of bits relative to the last frame. Bit-rates may vary, for example, in large predefined increments/decrements, or the bit-rates may vary by as little as one-bit resolution. The variability in bit rate may be either network controlled or source controlled according to the input audio signal.
Source controlled rate (SCR) devices use the source to vary the bit rate. For instance, during a normal telephone conversation, the participants alternate speaking so that, on average, each direction of transmission is occupied 50% of the time. SCR is a mode of operation where the speech encoder encodes speech frames containing only background noise with a lower speech rate than might be used for encoding speech. A network may also vary its transmission scheme to take advantage of the varying bit rate. Benefits provided therein include: (a) increases battery life and/or reduces power consumption of the associated processing system, and (b) the average required bit-rate is reduced, thereby leading to a more efficient transmission with decreased load and hence increased capacity.
Encoders conform to various standards proposed through the International Telecommunication Union (ITU) and European Telecommunications Standards Institute (ETSI). The ITU Telecommunication Standardization Sector (ITU-T) is one of the three Sectors of the ITU. The ETSI focuses moreso on European standards. The mission of the ITU and the ETSI is to ensure efficient production of high quality standards covering certain fields of telecommunications. Table 1 (ITU) and Table 2 (ETSI) show certain representative encoding standards:
TABLE 1ITU-T Standard EncodersNameAnnex (if appropriate)Approximate data rate(s)G.72816 Kbps/12.8 KbpsG.729A8 KbpsBVADD6.4 KbpsE12.8 KbpsG.71164 KbpsG.72616/24/32/40 Kbps  G.723.16.3 Kbps5.3 KbpsAVAD (voice activated device)
TABLE 2ETSI - Standard encodersGSM-AMR4–12KbpsGSM-FR/EFR13KbpsGSM-HR6.4Kbps
These representative encoders are part of a growing list of standards, with each different device being used according to its specification of abilities for a given situation. For instance, G.728 is an international voice compression standard from ITU and has rapidly gained acceptance for many applications including: satellite, cellular, and video-conferencing systems. G.728 is specified as part of the H.320 international video-conferencing standard. One reason for its rapid acceptance is that G.728 delivers the same toll-quality voice as 32 Kbps ADPCM (adaptive differential pulse-code modulation), but in only half the bandwidth. Note that ADPCM is a technique for converting sound or analog information to binary information (a string of 0's and 1's) by taking frequent samples of the sound and expressing the value of the sampled sound modulation in binary terms.
Referring to FIG. 6 of the above-incorporated references entitled “Voice and Data Exchange over a Packet Based Network with Resource Management,” and “Voice and Data Exchange over a Packet Based Network,” respectively, the encoders would be located within the voice encoder block 82. The encoders would be implemented via either hardware and/or software, depending upon the configuration that was implemented.
Certain members of this group of representative encoders are known to incorporate various aspects of SCR. For instance, the ETSI GSM-AMR coder is source controlled, as well as Qualcom's QCELP coder, and certain VAD-Speech coder combinations. Network management systems in current use will normally manage network bandwidth by downspeeding endpoints during congestion. However, the various members of this overall collection of encoders are not believed to use a collection (or combination) of statistical information—that might be derived from the network (i.e., the devices in the network and the associated connections) and the source—in order to arrive at a data rate decision. It is one important aspect to collectively consider the data rate needs of all of the devices in the network. It is also an important aspect to have a system that can adjust the rates of each device to thereby create a more efficient throughput of data across the entire system.
Accordingly, what is needed in the field is central controller coupled with a network of devices which can monitor statistical information (or the like) associated with each network device. The information might be considered alone or in combination with other information and thereby adjust the data rate of each device based upon such statistical information.